[OpenSER-Users] Calls disconnect automatically

[Iñaki Baz Castillo]

On Monday 21 January 2008 00:52:24 VoIP Forums www.Go4Calls.com wrote:
> The problem is only with PSTN call.
>
> I tried to send call to the three gateway Teles, SIP-HIT and Asterisk but
> all disconnect calls in that priticular seconds. The thinng is i cannot
> understand if i am using STUN in Linksyspap2 the call goes normal and
> without STUN it disconnect. So the problem is gateway side or Openser?
>
> our router is not implimented with SIP, and there is one more strange
> thing, In some callshop the same rtptproxy working well and going cal for
> long duration but i have 3 callshop which facing this problem. the
> configuration and others are same as other working devices.

Try the "tcpdump" I suggested in client side, you will discover when audio is 
cut.

-- 
Iñaki Baz Castillo
ibc at in.ilimit.es